similar to: Restricting outbound dialing on a specific phone

That sound's like the right thing to do, you'd probably have contexts related to what phones could do "unrestricted","localdialling","extensionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension

tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit > <salah.elharit200 at gmail.com> wrote: > >

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Hi, I am having a few dial plan problems which I wondered if anyone would be able to help with. Firstly, I wanted to send 0800 calls through 1 sip provider and other 08xx calls through another. I have this: exten => _0800.,1,Dial(SIP/${EXTEN}@provider1,30) exten => _0800.,2,Congestion exten => _08.,1,Dial(SIP/${EXTEN}@provider2,30) exten => _08.,2,Congestion However,

This seems basic but something is missing..... I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup application? Syntax issue somewhere? I have a good SIP registration with the vendor, voipvoip.

This page http://www.freecall.com/en/index.html is advertising free calls to: Argentina, Australia, Austria, Belgium, Canada, Czech Republic, Denmark, France, Germany, Hong Kong (+mobile), Hungary, Ireland, Italy, Luxembourg, Malaysia, Monaco, Netherlands, New Zealand, Norway, Panama, Poland, Portugal, Puerto Rico (+mobile), Russian Federation, Singapore, Slovenia, South Korea, Spain, Sweden,

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

Hi, I'm trying to mount samba shares onto the file system, but at the moment i only seem to be able to mount them as root. I've tried changing the permissions on the mount point and even mounting it on a subdir of my home directory but mount just keeps telling me that only root can do this. I've put an entry for the mount in /etc/fstab with the user option, what have i missed? I'm

Dear R list, I have the following data: set.seed(1) n <- 5 # number of subjects tt <- 3 # number of repeated observation per subject numco <- 2 # number of covariates x <- matrix(round(rnorm(n*numco),2), ncol=numco) # the actual covariates x > x [,1] [,2] [1,] -0.63 -0.82 [2,] 0.18 0.49 [3,] -0.84 0.74 [4,] 1.60 0.58 [5,] 0.33 -0.31 I need to form a matrix

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Hi I thought this might be of general interest. Recently I purchased an X100P from a Digium reseller in the UK. Very pleased with the card; works perfectly. My friend (known for his deep pockets and short arms) purchased an X100P card from Ebay. He's had no end of problems with line noise, dropped called etc so I thought I would compare the two cards. Plus mine was delivered within two

https://bugzilla.mindrot.org/show_bug.cgi?id=1586 Summary: [FEATURE REQ] PermitRootLogin and restricting to specific hosts Product: Portable OpenSSH Version: 5.2p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: Miscellaneous

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

I am using SSSD to get user AUTH from a backend Samba4 AD/DC. For Linux clients sssd.conf is configured to query Samba4 AD based on LDAP/Kerberos i.e. the Linux clients have not done a Domain join. Physical console logins -- things are working fine with changes to NSS and PAM (tool authconfig) for domain User AUTH on Linux and Windows clients. However, I want to restrict access to certain

I would think the /etc/hosts.allow and /etc/hosts.deny would do the trick. /etc/hosts.allow icecast: 127.0.0.1 64.147.130. 192.168.0. /etc/hosts.deny ALL: ALL Someone please correct me if I am wrong. It's been known to happen. KJ On Mon, 2003-02-24 at 09:55, Jesper Johnson wrote: > Hello > > I tried searching the mailing list and did some googling on the topic also, > but

Jesper Johnson expunged (jesper.johnson@welho.com): > I tried searching the mailing list and did some googling on the topic also, > but didn't find anything very useful. I'm looking for a way to restrict > access to my icecast2 server (which runs on a Slackware 8.1 system) > by using a list of IPs that should be allowed to connect as clients / sources. > .. now how do I

I have created a ''admin'' controller and all the necessary definitions and templates to delete, update and edit records and this is working OK, but how do I effectively restrict access to these defs'' from all but the admin user (called admin)? Currently I am just using if statements within the defs'' to check if it is the admin user logged in e.g; def index

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk